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"playout-delay mode fixed no-timestamps" What is it for

Performing an analysis after claim that the voice was robotic and poor quality, I found that removing the command "playout-delay mode fixed in the timestamps" within the dial-peer voice of my circuit the problem was normalized.
I was in doubt, why removing this command solved the problem, for it really serves this command.

 

dial-peer voice 525 voip

 description TESTE

 destination-pattern [2-6].......

 rtp payload-type cisco-codec-fax-ack 111

 rtp payload-type cisco-codec-fax-ind 110

 rtp payload-type nse 102

 rtp payload-type nte 97

 modem passthrough nse codec g711alaw

 voice-class codec 1

 session protocol sipv2

 session target sip-server

 incoming called-number .T

 dtmf-relay rtp-nte

 playout-delay mode fixed no-timestamps

 fax protocol t38 ls-redundancy 0 hs-redundancy 0 fallback pass-through g711alaw

 no vad

1 Reply 1

Playout delay is the amount of time that elapses between the time at which a voice packet is received at the jitter buffer on the DSP and the time at which it is played out to the codec. In most networks with normal jitter conditions, the defaults are adequate and you will not need to configure the playout-delaycommand.

In situations in which you want to improve voice quality by reducing jitter or you want to reduce network delay, you can configure playout-delay parameters. The parameters are slightly different for each of the two playout-delay modes, adaptive and fixed (see the playout-delay mode command).

In adaptive mode, the average delay for voice packets varies depending on the amount of interarrival variation that packets have as the call progresses. The jitter buffer grows and shrinks to compensate for jitter and to keep voice packets playing out smoothly, within the maximum and minimum limits that have been configured. The maximum limit establishes the highest value to which the adaptive delay will be set. The minimum limit is the low-end threshold for incoming packet delay that is created by the adaptive jitter buffer. Algorithms in the DSPs that control the growth and shrinkage of the jitter buffer are weighted toward the improvement of voice quality at the expense of network delay: jitter buffer size increases rapidly in response to spikes in network transmissions and decreases slowly in response to reduced congestion.

In fixed mode, the nominal value is the amount of playout delay applied at the beginning of a call by the jitter buffer in the gateway and is also the maximum size of the jitter buffer throughout the call.

As a general rule, if there is excessive breakup of voice due to jitter with the default playout-delay settings, increase playout-delay times. If your network is small and jitter is minimal, decrease playout-delay times for a smaller overall delay.

When there is bursty jitter in the network, voice quality can be degraded even though the jitter buffer is actually adjusting the playout delay correctly. The constant readjustment of playout delay to erratic network conditions causes voice quality problems that are usually alleviated by increasing the minimum playout-delay value in adaptive mode or by increasing the nominal delay for fixed mode.

 

playout-delay mode (dial peer)

To select fixed or adaptive mode for playout delay from the jitter buffer on digital signal processors (DSPs), use the playout-delay mode command in dial peer configuration mode. To reset to the default, use the no form of this command.

playout-delay mode {adaptive | fixed}

no playout-delay mode

Syntax Description

 
 

adaptive

 

Jitter buffer size and amount of playout delay are adjusted during a call, on the basis of current network conditions.

 

fixed

 

Jitter buffer size does not adjust during a call; a constant playout delay is added.

 

http://www.cisco.com/c/en/us/td/docs/ios/voice/command/reference/vr_book/vr_p2.html

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